CN102881281B

CN102881281B - Noise cancellation system with lower rate emulation

Info

Publication number: CN102881281B
Application number: CN201210277749.1A
Authority: CN
Inventors: A·J·马格拉思; R·克莱默
Original assignee: Wolfson Microelectronics PLC
Current assignee: Hongkong Semiconductor Co.,Ltd.; Cirrus Logic International UK Ltd; Cirrus Logic International Semiconductor Ltd
Priority date: 2007-12-21
Filing date: 2008-12-12
Publication date: 2015-04-08
Anticipated expiration: 2028-12-12
Also published as: GB0725111D0; US10147413B2; GB0810995D0; US20180247634A1; US20100310086A1; US20170213537A1; US20170213538A1; CN104751839B; CN104751839A; US8908876B2; CN101903941B; WO2009081187A1; CN101903941A; GB2455822A; GB2455822B; CN102881281A; US9654871B2; US10431198B2; US20150078569A1; US20170213539A1

Abstract

In a noise cancellation system comprising an input for a digital signal having a first sample rate and an adaptive digital filter 44 connected to the input, a decimator 52 is connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate. A processor 54 for controlling the adaptive digital filter 44includes an emulation 56 of the digital filter, connected to receive the decimated signal and to generate an emulated filter output.

Description

There is the noise canceling system of lower rate emulation

The application is the applying date is on Dec 12nd, 2008, application number is 200880122246.9(international application no PCT/GB2008/051182), name is called the divisional application of the application for a patent for invention of " noise canceling system with lower rate emulation ".

The present invention relates to noise canceling system, in particular to so a kind of noise canceling system: it has can the easily wave filter of adaptation based on input signal, to improve noise removing performance.

Background technology

Noise canceling system is known, and wherein, the electronic noise signal representing neighbourhood noise is applied to signal processing circuit, and then obtained treated noise signal is applied to loudspeaker, to generate voice signal.Eliminate to realize noise, the sound generated, with regard to its amplitude and its phase place, should be similar to the reversion thing (inverse) of neighbourhood noise as closely as possible.

Especially, feed-forward noise elimination system for head phone (headphone) or earphone (earphone) is known, wherein, one or more loudspeakers (microphone) ambient noise signal to wearer's ear region be arranged on head phone or earphone detects.Eliminate to realize noise, after neighbourhood noise self is by head phone or earphone correction, the sound generated needs the reversion thing being similar to this neighbourhood noise as far as possible.An example of the correction that head phone or earphone carry out be the noise of being advanced by the edge around head phone or earphone arrive wearer's ear must through different acoustic path cause.

In practice, be used for testing environment noise signal loudspeaker and be used for generating the loudspeaker (loudspeaker) of voice signal from treated noise signal and also will change these signals, such as compare sensitiveer in other frequency in some frequency.An example is: when loudspeaker is closely coupled to the ear of user, causes the frequency response of loudspeaker to change because of chamber effect (cavityeffect).

It is favourable for can carrying out adaptation (adapt) to the characteristic of the wave filter be used in signal processing circuit (characteristics), such as, in order to consider the character (properties) of neighbourhood noise.But when using high sampling rate, this filter adaptation can expend the power of appreciable amount.

Summary of the invention

According to a first aspect of the invention, provide a kind of noise canceling system, it comprises: for the input end (input) of digital signal, this digital signal has the first sampling rate; Digital filter, it is connected to this input end to receive this digital signal; Decimator (decimator), it is connected to this input end to receive this digital signal, and generates selection signal (decimated signal) with the second sampling rate lower than this first sampling rate; And processor.This processor comprises: the emulation (emulation) of this digital filter, and it is connected to receive this selection signal and generates emulated filter and exports (emulated filter output); And control circuit, it generates control signal based on exporting by this emulated filter, and wherein this control signal is applied to this digital filter to control its filter characteristic.

This has the following advantages: can control this digital filter based on this input signal, and generates control signal to be applied to this wave filter without the need to power intensive (power-intensive).

According to a second aspect of the invention, a kind of method eliminating neighbourhood noise is provided.The method comprises: receive digital signal, this digital signal has the first sampling rate; With digital filter, filtering is carried out to described signal; Generate selection signal from described digital signal, this selection signal has the second sampling rate lower than this first sampling rate; Use described selection signal to emulate this digital filter, thus generate emulated filter output; And, based on this emulated filter exports, control the filter characteristic of this digital filter.

Accompanying drawing explanation

For understanding the present invention better, and in order to more clearly show how can realize the present invention, now will in an illustrative manner with reference to the following drawings, wherein:

Fig. 1 illustrates noise canceling system according to an aspect of the present invention;

Fig. 2 illustrates in the noise canceling system of Fig. 1, according to an aspect of the present invention signal processing circuit;

Fig. 3 is a process flow diagram, which illustrates process (process) according to an aspect of the present invention;

Fig. 4 illustrates signal processing circuit according to the present invention and is implemented in feedback noise elimination system;

Fig. 5 illustrates in the noise canceling system of Fig. 1, according to an aspect of the present invention another signal processing circuit;

Fig. 6 shows the signal chart of applied gain with an embodiment of the change of the noise envelope detected;

Fig. 7 shows the signal chart of applied gain with another embodiment of the change of the noise envelope detected;

Fig. 8 illustrates in the noise canceling system of Fig. 1, according to a further aspect of the invention signal processing circuit;

Fig. 9 is a process flow diagram, which illustrates according to an aspect of the present invention, calibration noise canceling system method;

Figure 10 is a process flow diagram, which illustrates according to a further aspect of the invention, calibration noise canceling system method;

Figure 11 illustrates described with reference to Figure 8, realize in feedback noise elimination system according to signal processing circuit according to the present invention;

Figure 12 illustrate in the noise canceling system of Fig. 1, according to the signal processing circuit of another aspect of the invention; And

Figure 13 show according to an embodiment of the invention, gain is with the signal chart of the change of signal to noise ratio (S/N ratio).

Embodiment

Fig. 1 generally illustrates form according to audible spectrum noise canceling system of the present invention (audiospectrum noise cancellation system) and use.

Particularly, Fig. 1 shows earphone 10, and it is worn on the external ear 12 of user 14.Thus, Fig. 1 shows ear-sticking (supra-aural) earphone be worn on ear, but should recognize, identical principle is applicable to cover ear formula (circumauralheadphone) receiver worn around ear, and is worn over the earphone such as so-called In-Ear Headphones (ear-bud phone) in ear.The present invention is equally applicable to other and is intended to wear or remain on the equipment near user's ear, such as mobile phone, headset (headset) and other communication facilities.

Neighbourhood noise is detected by loudspeaker 20,22, and these two loudspeakers are shown in Figure 1, but also can be provided with any amount of loudspeaker greater or less than two.The ambient noise signal generated by loudspeaker 20,22 is combined, and is applied to signal processing circuit 24, and it will be described in more detail hereinafter.Loudspeaker 20,22 be simulation a micropkonic embodiment in, these ambient noise signals can combine by being added.When loudspeaker 20,22 is digital amplifiers, namely when their generate represent the digital signal of neighbourhood noise, these ambient noise signals can with other kind of Combination of Methods, as those of ordinary skill in the art are familiar with.Further, before these neighbourhood noises are combined, this loudspeaker can be applied in different gains, such as, in order to compensate the sensitivity difference caused because of manufacturing tolerance.

This shown embodiment of the present invention also comprises the source 26 of wanted signal.Such as, the entrance of the wanted signal that when being in use in---being such as intended to the earphone 10 that can regenerate music---at this noise canceling system at earphone, source 26 can be from external source---such as sound reproduction equipment such as MP3 player---connects (inlet connection).In other application, such as, when this noise canceling system is in use in mobile phone or other communication facilities, source 26 can comprise for received RF signal and the wireless receiver circuit of being decoded.In further embodiment, source can be there is no, and this noise canceling system can only be intended in order to the comfortable of user and eliminate neighbourhood noise.

From the wanted signal in source 26, if any, be applied to loudspeaker 28 by signal processing circuit 24, loudspeaker 28 nearby generates voice signal at the ear 12 of user.In addition, signal processing circuit 24 generted noise erasure signal, this noise-cancelling signal is also applied to loudspeaker 28.

An object of signal processing circuit 24 is generted noise erasure signals, this noise-cancelling signal is when being applied to loudspeaker 28, loudspeaker 28 is made to generate voice signal in the ear 12 of user, this voice signal is the reversion thing of the ambient noise signal arriving ear 12, is eliminated at least in part to make neighbourhood noise.

In order to realize this point, signal processing circuit 24 needs from the ambient noise signal generated by loudspeaker 20,22 to generate noise-cancelling signal, this noise-cancelling signal considers the performance of loudspeaker 20,22 and the performance of loudspeaker 28, and also considers that the neighbourhood noise caused because of the existence of earphone 10 changes.

Fig. 2 illustrates in greater detail the form of signal processing circuit 24.Input end 40 is connected to receive---such as direct receive from loudspeaker 20,22---input signal.This input signal is applied to analog to digital converter 42, and is converted into digital signal here.Then obtained digital signal is applied to can adaptive (adaptable) digital filter 44, and the signal through filtering obtained be applied to can adaptation device 46.

The output signal of adaptation device 46 can be applied to totalizer (adder) 48, here this output signal is added with the expectation source signal received from the second input end 49, and source 26 can be connected to the second input 49.Certainly, this is applicable to the embodiment that there is wanted signal.In the embodiment that there is not wanted signal (that is, this noise canceling system is designed to purely be used for reducing neighbourhood noise, such as, in high-noise environment), input end 49 and totalizer 48 are unnecessary.

Thus, the filtering that applies of wave filter 44 and gain apparatus 46 and horizontal adjustment (leveladjustment) are intended to generate the noise-cancelling signal that the neighbourhood noise that allows to detect is eliminated.

The output of totalizer 48 is applied to digital to analog converter 50, can be sent to loudspeaker 28 to make it.

As described above, this noise-cancelling signal is by can adaptive digital filter 44 and can producing from input signal by adaptation device 46.These are controlled by one or more control signal, this one or more control signal be by the digital signal exported from analog to digital converter 42 is applied to reduce digital sampling rate decimator 52, be then applied to microprocessor 54 and generate.

Microprocessor 54 comprises module (block) 56, it emulates wave filter 44 and gain apparatus 46 and produces emulated filter and exports, this emulated filter exports and is applied to totalizer 58, and here this emulated filter exports and is added with via the wanted signal of decimator 90 from the second input end 49.The sampling rate that decimator 52 performs reduces (sample ratereduction) and allows to perform this emulation with the power consumption lower than the emulation performed with original 2.4MHz sampling rate.

The signal obtained is applied to control module 60, and control module 60 generates the control signal of the performance for adjusting wave filter 44 and gain apparatus 46.By frequency bending (frequencywarping) module 62, smoothing filter (smoothing filter) 64 and sampling hold circuit 66, the control signal for wave filter 44 is applied to wave filter 44.Identical control signal is also applied to module 56, mates to make the adaptation of the emulation of wave filter 44 and wave filter 44 self.In one embodiment, for the control signal of wave filter 44 be device 58 with additive the comparing of output and a threshold value based on and generate.Such as, if the output of totalizer 58 is too high, then control module 60 can generate a control signal and reduces to make the output of wave filter 44.In one embodiment, this can be realized by the cutoff frequency reducing wave filter 44.

The object of frequency bending module 62 makes the control signal exported from control module 60 be adapted to high-frequency adaptation (adaptive) wave filter 82.That is, high frequency filter 82 usually can run with the frequency more much higher than the frequency of low frequency filter emulator 86, and therefore this control signal usually needs to be adapted and is applicable to this two wave filters with double.Therefore, the bending of this frequency can be replaced by any normal map function.

Any ripple (ripple) in the control signal that smoothing filter floating control module 60 generates, to make the noise reduction in this system.In an alternative embodiment, sampling hold circuit 66 can be replaced by interpolation filter (interpolation filter).

Control module 60 also generates the control signal being used for adaptive gain device 46.In the embodiment illustrated, gain control signal is output directly to gain apparatus 46.

In this preferred embodiment of the present invention, the digital signal being applied to this device is optionally oversampled (oversampled).That is, this digital signal sampling rate than process institute pay close attention to frequency range needs much higher times of nyquist frequency.But this higher sampling rate is combined with lower bit accuracy, to allow to process faster with accuracy high acceptably in digital filter 44.Such as, in one embodiment of the invention, the sampling rate of this digital signal is 2.4MHz.

But, find to run microprocessor 54 and wave filter emulation 56 with such high sampling rate.Thus, in the embodiment at this shown, sampling rate is reduced to 8kHz by decimator 52---and can be processed comfortably by microprocessor 54 and still keep sampling rate low in energy consumption.

Although Fig. 2 illustrates, first this control signal is applied to frequency bending module 62 and is then applied to smoothing filter 64, and the position of these modules can exchange.

Frequency bending module 62 is based on bilinear transformation, and this guarantees that emulating from low rate the control coefrficient obtained correctly is converted to the control coefrficient that must be applied to the wave filter 44 of high sampling rate operation, with the control realized a plan.

In this shown embodiment of the present invention, digital filter 44 comprises: fix level 80, and it takes the form of six rank iir filters, and its filter characteristic can be adjusted at calibration phase but after this keep fixing; And self-adaptation level 82, it takes the form of Hi-pass filter, its filter characteristic can in use based on input signal character and be adapted.Like this, the characteristic of digital filter 44 can be adapted based on neighbourhood noise.In one embodiment, this filter characteristic is the cutoff frequency of digital filter 44.

Therefore, also comprise the module 56 that digital filter 44 emulates: fix level 84, its filter characteristic can be adjusted at calibration phase but after this be kept fixing; And self-adaptation level 86, it takes the form of Hi-pass filter, its filter characteristic can in use based on input signal character and be adapted, especially based on control module 60 output and be adapted.

Although the fix level 80 of digital filter 44 is six rank iir filters, the fix level 84 of emulation 56 can be the iir filter of lower-order, such as second order IIR filter, and this still can provide and emulates accurately acceptably.

Further, microprocessor 54 can comprise an adaptive gain emulator (not shown in Fig. 2), and it is between wave filter emulator 56 and totalizer 58.In the case, control module 60 also will output to this adaptive gain emulator gain control signal.

Under the prerequisite of scope not departing from this instructions claims, various amendment can be made to above-described embodiment.Such as, the source signal being input to signal processor 24 can be digital, as described above, or simulation---and analog to digital converter may be necessary to convert this signal to numeral in this case.Further, can select digital source signal in decimation filter (not shown).

As mentioned above, the digital signal representing the neighbourhood noise detected is applied to adaptive digital filter 44, with generted noise erasure signal.In order to signal processing circuit 24 can be used in a variety of different applications, adaptive digital filter 44 is necessary relative complex, to make it can compensate for different loudspeakers and speaker combination, and for there being the dissimilar earphone of Different Effects to compensate to neighbourhood noise.

But, must---such as iir filter---execution complete adaptive (full adaptation) can be disadvantageous to the wave filter of complexity in the use of equipment.Thus, in this preferred embodiment of the present invention, wave filter 44 comprises such iir filter 80: its filter characteristic is fixed effectively when this equipment is in running.More specifically, this iir filter can have some groups of possible filter coefficients, these filter coefficients are limiting filtering device characteristic together, loudspeaker 20 and 22, loudspeaker 28 and earphone 10 that one of these groups of filter coefficients are just using based on signal processing circuit 24 and be applied in.

When the arranging of iir filter coefficient can occur in and manufacture this equipment, or occur in when first this equipment being inserted specific earphone 10, or as the result of the calibration process occurred when this equipment initial power-up or with periodic intervals (such as once a day).After this, filter coefficient no longer changes, thus filter characteristic is fixed, but not changes based on applied signal.

But, find that this may have following shortcoming: this equipment may can not work under all conditions all best.Such as, when there being relatively high low-frequency noise level, the noise-cancelling signal obtained can be in the level of higher level more treatable than general loudspeaker 28.

Thus, wave filter 44 also comprises self-adaptive component, is self-adaptation Hi-pass filter 82 in the embodiment at this shown.Adjusted based on the control signal that---such as cutoff frequency---can generate by microprocessor 54 so the performance of this Hi-pass filter.And the adaptation of wave filter 44 can occur based on control signal simply too much.

Therefore, use the wave filter comprising fixed part and self-adaptation part to allow for the wave filter using relative complex, but also allow for by means of next this wave filter adaptive of relatively simple control signal.

As described so far, the adaptation of wave filter 44 is to occur based on the control signal obtained to the input of this wave filter.But also possible that, the adaptation of wave filter 44 can occur to export from this wave filter based on the control signal obtained.And, this wave filter is divided into fixed part and self-adaptation part allow for following possibility: the adaptation of wave filter 44 can occur based on the control signal obtained from the output of the first order in these filter stages.Especially, as shown, when this signal be first applied to the first fix level 80 be then applied to sef-adapting filter level 82, the adaptation of sef-adapting filter level 82 can occur based on the control signal obtained from the output of the first fix level 80.

As described above, this control signal is by microprocessor 54---it comprises the emulation of wave filter 44---generates.Therefore, when wave filter 44 comprises fix level 80 and self-adaptation level 82, emulation 56 preferably should also comprise fix level 84 and self-adaptation level 86, can be adapted in an identical manner to make it.

In this shown embodiment of the present invention, wave filter 44 comprises fixing iir filter 80 and self-adaptation Hi-pass filter 82, and similarly, wave filter emulation 56 comprises fixing iir filter 84 and self-adaptation Hi-pass filter 86, and fixing iir filter 84 and self-adaptation Hi-pass filter 86 are the reflections (mirror) of the wave filter that they emulate or are enough similar to accurately.

But the present invention can be applied to any following filter arrangement: wherein this wave filter comprises one or more filter stage, as long as at least one such level is adaptive.Such as, and this wave filter can relative complex, is iir filter, or can be relatively simple, be such as low order low pass or Hi-pass filter.

Further, possible filter adaptation can relative complex, and some different parameters are adaptive, or can be relatively simple, only have a parameter to be adaptive.Such as, in the embodiment at this shown, self-adaptation Hi-pass filter 82 is by the controllable firstorder filter of single controlling value, and this controlling value has the effect changing wave filter corner frequency (corner frequency).But in other situations, this adaptation can take the form of the some parameters changing higher order filters, or can take the form of a complete set of filter coefficient changing iir filter in principle.

As everyone knows, in order to processing digital signal, be necessary that the signal of the sampling rate of at least twice by the frequency with this signal message content operates, and the component of signal being in the frequency of the half higher than this sampling rate will be lost.When being in the high signal to the frequency of cutoff frequency and must being processed, define nyquist sampling rate, it is the twice of this cutoff frequency.

Noise canceling system is intended to only eliminate usually can listen effect (audible effects).Because the upper limiting frequency (upper frequency) of human auditory is generally 20kHz, which imply that and can realize acceptable performance by carrying out sampling with the sampling rate of about 40kHz to noise signal.But, in order to realize enough performances, the degree of accuracy required with relatively high is sampled to noise signal, and inevitably will there is delay in the process to such signal.

Therefore, in this shown embodiment of the present invention, analog to digital converter 42 generates digital signal with the sampling rate of 2.4MHz, but has the bit resolution (bitresolution) of only 3 bits.This allow that signal transacting accurately acceptably, but there is much lower signal transacting postpone.In other embodiments of the present invention, the sampling rate of this digital signal can be 44.1kHz, or is greater than 100kHz, or is greater than 300kHz, or is greater than 1MHz.

As described above, wave filter 44 is adaptive.That is, can transmit control signal to change its performance to this wave filter, such as its frequency characteristic.In this shown embodiment of the present invention, this control signal is not send with the sampling rate of this digital signal, but send with lower speed (rate).Process complexity during to which save power and this control circuit---be microprocessor 54---in the case.

This control signal is sent out with such speed: this speed allows enough this wave filters adaptive rapidly of this control signal may produce with process the change can listening effect, that is, this speed at least equals the nyquist sampling rate that defined by the expectation cutoff frequency in audiorange.

Although expect that can realize noise in whole audiorange eliminates, and in practice, only likely realizes good noise removing performance usually in a part for audiorange.In the ordinary course of things, be considered to preferably: optimize this system,---in the scope such as from 80Hz to 2.5kHz---to realize good noise removing performance in the comparatively lower part of audiorange.Therefore, it is exactly enough for generating the control signal with following sampling rate: this sampling rate is the twice of following frequency, does not expect to realize remarkable noise removing performance more than this frequency.

In this shown embodiment of the present invention, this control signal has the sampling rate of 8kHz, but, in other embodiments of the present invention, this control signal can have and is less than 2kHz, or is less than 10kHz, or be less than 20kHz, or be less than the sampling rate of 50kHz.

In this shown embodiment of the present invention, the sampling rate of this digital signal is reduced to 8kHz from 2.4MHz by decimator 52, and microprocessor 54 produces control signal with the sampling rate identical with its input signal.But microprocessor 54 can produce the control signal with following sampling rate in principle: input signal that this sampling rate ratio receives from decimator 52, this microprocessor is higher or lower.

This shown embodiment shows, and this noise signal by---such as loudspeaker---receives from dummy source, and to be converted into digital form in the analog to digital converter 42 of this signal processing circuit.But, it will be appreciated that this noise signal is passable, such as, from digital amplifier, be received in digital form.

Further, this shown embodiment shows, and this noise-cancelling signal is generated in digital form, and is converted into analog form in the digital to analog converter 50 of this signal processing circuit.But, it will be appreciated that this noise-cancelling signal can be output in digital form, such as, for being applied to digital loudspeaker or analog.

In one embodiment of the invention, iir filter 80 has following filter characteristic: the signal preferably making to be in relatively low frequency passes through.Such as, although this noise canceling system may be attempted to eliminate neighbourhood noise as much as possible on whole audio band, but the specific arrangements of loudspeaker 20,22 and loudspeaker 28, and the size and dimension of earphone 10 may mean, for iir filter 80 preferably, there is the filter characteristic that lifting (boost) is in the signal of the frequency in the scope of 250-750Hz.But in further embodiment, iir filter 80 can also have remarkable lifting at below 250Hz.This lifting may be needs for compensating the little loudspeaker---it has bad LF-response usually---be arranged in little shell.

But, this means, when existence has the ambient noise signal of large component in this frequency range, there is such danger: the noise signal that wave filter 80 generates may be greater than loudspeaker 28 can be abundant---undistorted ground etc.---noise signal of process, namely loudspeaker 28 may be overdriven.Just in case there is this situation, loud speaker (cone) may exceed it and depart from the limit (excursion limit), thus causes the physical damage of loudspeaker.

Therefore, in order to prevent this point, the frequency characteristic of self-adaptation Hi-pass filter 82 based on input signal amplitude and be adapted.In fact, in the preferred embodiment, the frequency characteristic of self-adaptation Hi-pass filter 82 is adapted based on the output signal from emulated filter 56.And in the preferred embodiment, the frequency characteristic of self-adaptation Hi-pass filter 82 is adapted based on the wanted signal from the second input end 49 and the output signal sum from emulated filter 56.This means, the frequency characteristic of self-adaptation Hi-pass filter 82 is adapted based on the representative of the signal that in fact can be applied to loudspeaker 28.

More specifically, in this shown embodiment of the present invention, self-adaptation Hi-pass filter 82 is single order Hi-pass filters, and its cutoff frequency or corner frequency can be adjusted based on the control signal applied from microprocessor 54.The frequency of wave filter 82 more than corner frequency has the gain of somewhat constant, this gain can be unit value (unity) or can be certain other value, as long as there is suitable compensation in the other places of this filter paths, and this wave filter 82 has the gain of reduction below this corner frequency.

In one embodiment, this corner frequency can be adjustable in the scope of 10Hz-1.4kHz.

Fig. 3 is a process flow diagram, which illustrates the process performed in control module 60.

In step 90, by arranging initial value for controlling value K by this procedure initialization, this controlling value K is used to the corner frequency controlling Hi-pass filter 82.

In step 92, to the input value of control module 60,---namely emulated filter module 56 and expectation source input the absolute value of 49 sum H---compares with threshold value T-phase.If should exceed threshold value T with H, then this process had forwarded step 94 to, in step 94, attacked coefficient (attackcoefficient) K _abe added to current control value K.After these values are added together, test new controlling value in step 96 and whether exceed higher limit, if so, then change into and apply this higher limit.If this new controlling value does not exceed this higher limit, then use the controlling value that this is new.

If in step 92, should with the absolute value of H lower than threshold value T, then this process forwards step 98 to, in step 98, decay coefficient K _dbe added to current control value K.It should be noted that decay coefficient K _dbe negative, therefore it added to current control value K and reduce this controlling value.After these values are added together, test new controlling value in step 100 and whether drop to below lower limit, if so, then change into and apply this lower limit.If this new controlling value does not drop to below this lower limit, then use the controlling value that this is new.

When determining new controlling value, this process gets back to step 92, and in step 92, the new and H that emulated filter module 56 and expectation source input 49 is compared by with threshold value T-phase.

In one embodiment, COEFFICIENT K is attacked _avalue is greater than decay coefficient K _d, to make if there is instantaneous low frequency signal, then to improve cutoff frequency rapidly, thus cause output amplitude to reduce fast, exceed it to prevent loudspeaker and depart from the limit.Further, relatively little attenuation coefficient makes any corrugated minimum of cutoff frequency, effectively follows the tracks of the envelope of input signal and not absolute terms to make cutoff frequency.

Further, it will be apparent to those of ordinary skill in the art that to suitably change cutoff frequency to prevent speaker overload, other realization of the control algolithm performed in control module 60 is possible.Such as, COEFFICIENT K is attacked _aand decay coefficient K _dcan change in non-linear (such as, index) mode.

As described above, this control procedure performs with the sampling rate lower than the sampling rate of supplied with digital signal.In order to ensure the root that this is not error, this controlling value is transmitted through frequency Warping function 62.

Further, this controlling value is transmitted through smoothing filter 64, and smoothing filter 64 is provided to any less desirable ripple in this signal floating.In this embodiment, this wave filter determines that this controlling value is in increase or in reduction.If this controlling value is in increase, then the output of wave filter 64 directly follows the tracks of input, is not with any time lag.But if this controlling value is in reduction, then the output of wave filter 64 decays towards input, exponentially with any less desirable ripple in floating output signal.

The output of smoothing filter 64 is passed to sampling hold circuit 66, and this output is from being drawn (latch out) here to sef-adapting filter 82.Then, by being applied to the controlling value through filtering of wave filter 82 to determine the corner frequency of this wave filter.Such as, when this controlling value takes lower limit, this corner frequency can take its minimum value---be 10Hz in the embodiment at this shown, and when this controlling value takes higher limit, this corner frequency can take its maximal value---be 1.4kHz in the embodiment at this shown.

It will be apparent to those of ordinary skill in the art that the present invention is equally applicable to so-called feedback noise and eliminates system.

Feedback method based on, the loudspeaker being placed on loudspeaker dead ahead being formed at the use in the chamber between ear and ear casing inside or between ear and mobile phone.Derive from micropkonic signal to be coupled via feedback loop (inverting amplifier) and to get back to loudspeaker, to make it form a servo-drive system, in this servo-drive system, loudspeaker is always attempted to produce zero sound pressure level (null sound pressure level) at loudspeaker place.

Fig. 4 shows and is implemented in feedback system according to the embodiment of signal processing circuit of the present invention.

This feedback system comprises the loudspeaker 120 generally within loudspeaker 128 front.Loudspeaker 120 detects the output of loudspeaker 128, and the signal detected is presented back via amplifier 141 and analog to digital converter 142.Expect that sound signal is fed into this treatment circuit via input end 140.In subtraction element (subtracting element) 188, from this expectation sound signal, deduct this by the signal presented back, to make the output of subtraction element 188 substantially represent neighbourhood noise, that is, expect that sound signal is eliminated substantially.

After this, this treatment circuit is similar to the treatment circuit 24 in the feedforward system described with reference to Fig. 2 substantially.The output of subtraction element 188 is fed into adaptive digital filter 144, and the signal through filtering is applied to can adaptation device 146.

The signal obtained is applied to totalizer 148, and here this signal is added with the expectation sound signal received from input end 140.

Thus, wave filter 144 and the filtering applied by gain apparatus 146 and horizontal adjustment are intended to generate the noise-cancelling signal that the neighbourhood noise that allows to detect is eliminated.

The output of totalizer 148 is applied to digital to analog converter 150, can be sent to loudspeaker 128 to make it.

As described above, this noise-cancelling signal is by adaptive digital filter 144 with can produce from input signal by adaptation device 146.These are controlled by a control signal, this control signal be by the digital signal exported from analog to digital converter 142 is applied to reduce digital sampling rate decimator 152, be then applied to microprocessor 154 and generate.

Microprocessor 154 comprises module 156, module 156 pairs of wave filters 144 and gain apparatus 146 emulate and produce emulated filter and export, this emulated filter exports and is applied to totalizer 158, and here this emulated filter exports and is added with via the expectation sound signal of decimator 190 from input end 140.

The signal obtained is applied to control module 160, and control module 160 generates the control signal of the performance for adjusting wave filter 144 and gain apparatus 146.By frequency bending module 162, smoothing filter 164 and sampling hold circuit 166, the control signal for wave filter 144 is applied to wave filter 144.Identical control signal is also applied to module 156, mates to make the adaptation of the emulation of wave filter 144 and wave filter 144 self.

In an alternative embodiment, sampling hold circuit 166 is replaced by interpolation filter.

Control module 160 also generates the control signal being used for adaptive gain device 146.In embodiment at this shown, this gain control signal is output directly to gain apparatus 146.

Further, microprocessor 154 can comprise the adaptive gain emulator (not shown in Fig. 3) between wave filter emulator 156 and totalizer 158.In the case, this gain control signal is also outputted to this adaptive gain emulator by control module 160.

Be similar to feedforward situation, fixed filters 180 can be iir filter, and sef-adapting filter 182 can be Hi-pass filter.

According to a further aspect in the invention, signal processor 24 comprises such device, and it is for measure ambient noise level and for carrying out the interpolation of control noises erasure signal to source signal based on ambient noise level.Such as, in neighbourhood noise in low or insignificant environment, this noise eliminates the sound quality may can not improved user and hear.That is, noise is eliminated and even artefact (artefacts) may be added into acoustic streaming (sound stream) to correct non-existent neighbourhood noise.Further, the activity of this noise canceling system within such period wastes power.Therefore, when noise signal is low, can reduce to turn off (turn off) noise-cancelling signal even completely.Which save power, and prevent noise signal that less desirable noise is added into voice signal.

But when this noise canceling system is present in such as mobile phone or headset (headset), neighbourhood noise can be isolated to voice that user controls oneself and detect.That is, user may speak facing to mobile phone or headset without in the room of his thing, but this noise canceling system still can not detect that because of the voice of user noise is low.

Fig. 5 illustrates in greater detail the another embodiment of signal processing circuit 24.Input end 40 is connected to receive---such as direct receive from loudspeaker 20,22---represents the noise signal of neighbourhood noise.This noise signal is imported into analog to digital converter (ADC) 42, and is converted into digital noise signal.This digital noise signal is imported into noise cancellation module 44, noise cancellation module 44 output noise erasure signal.Noise cancellation module 44 can such as comprise for the wave filter from the ambient noise signal generted noise erasure signal detected, that is, noise cancellation module 44 generates the reverse signal of the neighbourhood noise detected substantially.This wave filter can be adaptive or non-self-adapting, as obvious to those of ordinary skill in the art.

This noise-cancelling signal is output to variable gain module 46.The control of variable gain module 46 will be described after a while.Routinely, gain module can apply gain to noise-cancelling signal, to generate the noise-cancelling signal eliminating the neighbourhood noise detected more accurately.Thus, noise cancellation module 44 generally can comprise the gain module (not shown) being designed to operate in this way.But according to one embodiment of the invention, the gain applied changes according to the amplitude of the neighbourhood noise detected or envelope.Therefore, variable gain module 46 can be present in noise cancellation module 44 together with conventional gain module, or, be suitable for realizing the gain module that variable gain module 46 of the present invention itself can replace in noise cancellation module 44.

Signal processor 24 also comprises the input end 48 for receiving voice or other wanted signal, as described above.Thus, in the case of a mobile phone, wanted signal is the signal being transferred to this phone and having treated to convert to by means of loudspeaker 28 sub-audible sound (audible sound).Usually, this wanted signal can be digital (such as, music, the voice etc. received), and in the case, this wanted signal is added into the noise-cancelling signal exported from variable gain module 46 in adding element (adding element) 52.But when wanted signal is simulation, it is not shown that wanted signal is imported into ADC(), be converted into digital signal here, be then added in adding element 52.Then, composite signal is outputted to loudspeaker 28 by from signal processor 24.

Further, according to the present invention, digital noise signal is imported into envelope detector 54, the envelope of envelope detector 54 testing environment noise, and control signal is outputted to variable gain module 46.Fig. 6 shows an embodiment, and wherein envelope detector 54 is by the envelope of noise signal and threshold value N ₁compare, and compare output control signal based on this.Such as, if the envelope of noise signal is at threshold value N ₁below, then envelope detector 54 can export the control signal that zero gain is applied in, thus the noise cancellation of shutdown system 10 effectively.Similarly, envelope detector 54 can export control signal with the noise cancellation of in fact shutdown system 10.In the embodiment illustrated, if the envelope of noise signal is at first threshold N ₁below, then envelope detector 54 exports the control signal that gain is reduced gradually along with noise decrescence, makes when reaching second, lower threshold value N ₂time zero gain be applied in.At threshold value N ₁and N ₂between, gain is changed linearly; But, persons of ordinary skill in the art will recognize that this gain can such as stepwise or exponentially be changed.

Fig. 7 shows the schematic diagram of another embodiment, and wherein envelope detector 54 uses first threshold N by this way ₁with Second Threshold N ₂: delayed (hysteresis) is established in this system.The solid line of this figure represents the gain applied when this system is converted to zero noise-cancelling signal from " entirely " noise-cancelling signal; Dot-and-dash line (chain line) represents the gain applied when this system is converted to full noise-cancelling signal from zero noise-cancelling signal.In the embodiment illustrated, when this system initially generates full noise-cancelling signal, but neighbourhood noise drops to first threshold N afterwards ₁time following, the gain applied is reduced, until at neighbourhood noise value N ₁' place applies zero gain.When this system is initially turn-off state or generation " zero " noise-cancelling signal, and the envelope of neighbourhood noise rises to Second Threshold N ₂time above, the gain applied is increased, until at neighbourhood noise value N ₂' place generates full noise-cancelling signal.This Second Threshold can be set to higher than value N ₁'---eliminate at this value place noise and be previously switched off, make delayed being established in this system.This is delayed prevents the envelope when noise signal from eliminating rapid fluctuations between "ON" and "Off" state close to noise during this first threshold.

Persons of ordinary skill in the art will recognize that can turn off or open noise eliminates when neighbourhood noise crosses over the first and second threshold values respectively, but not reduce gradually or increase the gain applied.But in this embodiment, the envelope detector 54 of signal processor 24 can comprise ramp filter (ramping filter) and become level and smooth to make the transformation between different gains level.Sharply (harsh) transformation can sound strange to user, and by selecting reasonable time constant to avoid drastic shift for ramp filter.

Although use envelope detector to determine ambient noise level in the description above, the amplitude of noise signal also can be used to determine ambient noise level.Term " noise level "---it is also used in this manual---is applicable to the amplitude of noise signal or envelope or other value.

Certainly, there is many not specifically mentioned, significantly possible to those of ordinary skill in the art alternative methods here, to change the interpolation of noise-cancelling signal to wanted signal according to the neighbourhood noise detected.Except limiting in the following claims, the invention is not restricted to any one in described method.

According to another embodiment of the present invention, the digital noise signal exported from ADC 42 is imported into envelope detector 52 via door (gate) 56.Door 56 is controlled by voice activity detector (VAD) 58, and VAD 58 also receives the digital noise signal exported from ADC 42.Then, VAD 58 opposite house 56 operates, and makes noise signal only without being allowed to sensible envelope detector 52 in speech period.The operation of door 56 and VAD 58 will hereafter describe in more detail.When noise canceling system 10 is implemented in mobile phone or headset,---namely user tends to any system of speaking in use---is time middle, VAD 58 and door 56 especially useful.

Voice activity detector is used to be favourable, because this system comprises one or more loudspeaker 20,22, these loudspeaker testing environment noises, but also enough close to detect the speech of user oneself.Determine should control the gain of this noise canceling system based on neighbourhood noise time, can be favourable at testing environment noise level in user dumb period.

In this shown embodiment of the present invention, the noise level in the quietest period among the long term is taken as ambient noise level.Thus, in one embodiment---the signal wherein from loudspeaker 20,22 is become digital signal by the sample rate conversion with 8kHz, and these numeral samples are divided into some frames, and each frame comprises 256 samples, and is each frame determination average signal value.Then, the ambient noise level in any moment is confirmed as the frame among nearest 32 frames with minimum average signal value.

Thus, suppose have a frame to be that user does not manufacture any sound within the period (=about 1 second) of every section of 32 × 256 samples, then the signal level detected in this image duration will represent neighbourhood noise exactly.

Then, control based on the ambient noise level determined by this way the gain being applied to noise-cancelling signal.But certain, known many methods for detecting speech activity, thus, except this instructions is defined in the appended claims, the invention is not restricted to any ad hoc approach.

Under the prerequisite of scope not departing from this instructions claims, various amendment can be carried out to above-described embodiment.Such as, digital noise signal can be directly inputted to signal processor 28, in this case, signal processor 28 will not comprise ADC 42.Further, VAD 58 can receive the noise signal of analog form, but not digital signal.

The present invention can be used in feed-forward noise elimination system, as described above, or is used in so-called feedback noise elimination system.For these two kinds of systems, the rule of according to the ambient noise level detected noise-cancelling signal being added to wanted signal is all applicable.

Fig. 8 illustrates in greater detail the another embodiment of signal processing circuit 24.Input end 40 is connected to receive---such as direct receive from loudspeaker 20,22---input signal.This input signal is exaggerated in amplifier 41, and be applied to analog to digital converter 42 through amplifying signal, here this is converted into digital signal through amplifying signal.This digital signal is applied to adaptive digital filter 44, and the signal through filtering is applied to can adaptation device 46.Persons of ordinary skill in the art will recognize that at loudspeaker 20,22 are digital amplifiers---wherein analog to digital converter to be included in loudspeaker case (capsule) and input end 40 receives digital input signals---, do not need analog to digital converter 42.

The signal obtained is applied to the first input end of totalizer 48, and its output is applied to digital to analog converter 50.The output of digital to analog converter 50 is applied to the first input end of second adder 56, and the second input end of second adder 56 receives the wanted signal from source 26.The output of second adder 56 is sent to loudspeaker 28.Those of ordinary skill in the art also will recognize, this wanted signal can be imported into this system in digital form.In the case, before totalizer 56 can be positioned at digital to analog converter 50, the composite signal thus exported from totalizer 56 was converted into simulation before being exported by loudspeaker 28.

Thus, the filtering that applies of wave filter 44 and gain apparatus 46 and horizontal adjustment are intended to generate the noise-cancelling signal that the neighbourhood noise that allows to detect is eliminated.

As described above, this noise-cancelling signal is produced from input signal by adaptive digital filter 44 and adaptive gain device 46.These are controlled by a control signal, this control signal be by the digital signal exported from analog to digital converter 42 is applied to reduce digital sampling rate decimator 52, be then applied to microprocessor 54 and generate.

In this shown embodiment of the present invention, sef-adapting filter 44 is made up of the first filter stage 80 of fixing iir filter 80 form and the second filter stage of self-adaptation Hi-pass filter 82 form.

Microprocessor 54 generates a control signal, and this control signal is applied to self-adaptation Hi-pass filter 82 to adjust its corner frequency.In the use of this noise canceling system, microprocessor 54 generates this control signal on self-adaptation basis, can be adjusted to make the performance of wave filter 44 based on the character of the noise signal detected.

But the present invention is equally applicable to the system with fixing (fixed) wave filter 44.In this linguistic context, word " is fixed " and is meant, and the characteristic of this wave filter is not adjusted based on the noise signal detected.

But, the characteristic of wave filter 44 can be adjusted at calibration phase, when this calibration phase such as can occur in manufacturing system 24, or when first this system 24 being integrated in off-the-shelf hardware together with loudspeaker 20,22 and loudspeaker 28, or when this system is powered, or occur in the mode of other irregular (irregular).

More specifically, by being sent to wave filter 80 by replacing filter coefficient (a replacement set of filter coefficients) from a group in the many groups coefficient be stored in storer 90, can in the characteristic of the fixing iir filter 80 of this calibration phase adjustment.

Further, can similarly in the adjustment of this calibration phase by can the gain that applies of adaptation element 46.Or, by adjusting the characteristic of fixing iir filter 80 aptly, the change of this gain can be realized at this calibration phase.

Like this, just can for the concrete equipment treating to use together to optimize signal processing circuit 24.

Fig. 9 is a process flow diagram, which illustrates method according to an aspect of the present invention.As described above, this signal processing circuit needs to generate following noise-cancelling signal, and this noise-cancelling signal, when being applied to loudspeaker 28, produces the sound eliminating the neighbourhood noise that user hears as much as possible.The amplitude producing the noise-cancelling signal of this effect is by the sensitivity of the sensitivity and loudspeaker 28 of depending on loudspeaker 20,22, and depend on from loudspeaker 28 to the degree of coupling of loudspeaker 20,22 (such as, does loudspeaker 28 have many close to loudspeaker 20,22?), but can suppose that this is identical for all same model equipment (such as mobile phone).The method is set about from following understanding: although these two parameters are not easy to measure, real importantly their product.Therefore, the method according to this invention comprises: applying has the test signal of known amplitude to loudspeaker 28; And detect with loudspeaker 20,22 sound obtained.The amplitude of the signal detected is measuring of the product of the sensitivity of loudspeaker 20,22 and the sensitivity of loudspeaker 28.

In step 110, in microprocessor 54, a test signal is generated.In one embodiment of the invention, this test signal is the digitized representation thing of the sinusoidal signal being in given frequency.As described above, the object of this calibration process is the difference between compensation equipment, even if these equipment marks deserve to be called identical.Such as, in mobile phone or similar devices, micropkonic gain may or little 3dB larger than its nominal value.Similarly, the gain of loudspeaker may or little 3dB larger than its nominal value, and the product of both results may or little 6dB larger than its nominal value.In addition, loudspeaker generally can have resonance frequency in the somewhere in audio frequency range.Should recognize, if measurement be carry out in the resonance frequency of this loudspeaker and another measurement carries out away from the resonance frequency of that loudspeaker, then the result that measurement can provide misleading is carried out to relative (relative) gain of two loudspeakers, and, if these two loudspeakers have different resonance frequencies, even if then carry out gain measurement under same frequency, this situation also may occur.

Therefore, this test signal preferably includes the digitized representation thing of the sinusoidal signal being in given frequency, wherein this given frequency is rather away from any expection resonance frequency of this loudspeaker, therefore make all same categories of device all be expected and there is roughly similar character, except the roughly sensitivity of their loudspeaker and loudspeaker.

In some alternate embodiment, this test signal can be band-limited noise signal (band-limited noise signal), or pseudo-random data pattern (pseudo-random data-pattern), such as maximal-length sequence.

In step 112, this test signal is applied to the second input end of totalizer 48 from microprocessor 54, is thus applied to loudspeaker 28.

In step 114, loudspeaker 20,22 detects the voice signal obtained, and a part for the signal detected is sent to microprocessor 54.

In step 116, the amplitude of the signal detected measured by microprocessor 54.This can complete in a different manner.Such as, the net amplitude of the signal detected can be measured, but this will cause test sound not only being detected but also any neighbourhood noise being detected.Or, filtering can be carried out to the voice signal detected, and detect the amplitude through the voice signal of filtering.Such as, the voice signal detected can be conveyed through digital Fourier transform, thus allow by this voice signal, the component of the frequency that is in test signal separates, and measures its amplitude.As another replacement scheme, this test signal can comprise data pattern, and microprocessor 54 can be used to detect the correlativity (correlation) between detected voice signal and test signal, to make it possible to determine, the amplitude detected comes from this test signal but not comes from neighbourhood noise.

In step 118, this signal processor is adapted based on the amplitude detected.Such as, the gain of adaptive gain element 46 can be adjusted.

Signal processing circuit 24 is intended to be used in various equipment.But expectation can manufacture the equipment comprising signal processing circuit 24 in a large number, they are all included in the comparatively large equipment comprising loudspeaker 20,22 and loudspeaker 28.Although these can be identical compared with in large equipment nominal, each loudspeaker and each loudspeaker may slightly differences.The present invention sets about from following understanding: these differences, more significant one is, the difference of the resonance frequency of the loudspeaker 28 of each equipment.The present invention also sets about from following understanding: the resonance frequency of loudspeaker 28 can change along with the change of loudspeaker voice coil temperature the use of equipment.But other reason that resonance frequency changes also is possible, comprises aging or humidity change etc.The present invention is equally applicable to all such situations.

Figure 10 is a process flow diagram, which illustrates according to method of the present invention.In step 132, microprocessor 54 generates a test signal, and this test signal is applied to the second input end of totalizer 48.In one embodiment, this test signal is the sinusoidal signal that a succession of (aconcatenation of) is in multiple frequency.The frequency range residing for resonance frequency expection of these frequency coverages loudspeaker 28.

In step 134, the impedance of this loudspeaker is determined.That is, based on applied test signal, the electric current flowing through loudspeaker voice coil is measured.Such as, the electric current in loudspeaker voice coil can be detected, and is sent to microprocessor 54 through analog to digital converter 57 and decimator 59.Expediently, this microprocessor is passable, by the current signal detected being applied to digital Fourier transform module (not shown) and at the value of each frequency measurement current waveform, determining the impedance at each frequency place.Or the speed that can generate sample by suitably adjusting decimator 59 detects the signal being in different frequency.

In the step 136 of this process, determine resonance frequency, it is: across in the frequency band of (span) possible resonant frequency range, frequency that the minimum thus impedance of electric current is maximum.

In step 138, the frequency characteristic of wave filter 44 is adjusted based on the resonance frequency detected.In one embodiment, storer 90 stores organizes filter coefficient more, often organizes filter coefficient and defines the iir filter that has following characteristic: it comprises the peak value being in characteristic frequency.These characteristic frequency can be advantageously identical with the frequency of the sinusoidal signal forming this test signal.In the case, advantageously, apply to adaptive iir filter the coefficient that a group defines following wave filter: this wave filter has the peak value being in the resonance frequency detected.

In one embodiment of the invention, these group filter coefficients define six rank wave filters separately, and between these filter characteristics, the most essential difference is their resonance frequency.

Thus, likely detect the resonance frequency of this loudspeaker, and select the wave filter with the characteristic of mating most with it.

In embodiments of the invention, microprocessor 54 can comprise the emulation of wave filter 44, to allow to carry out adaptation based on the noise signal detected to the filter characteristic of wave filter 44.In the case, any filter characteristic being applied to wave filter 44 preferably also should be applied to the wave filter emulation in microprocessor 54.

Up to the present, the present invention is described with reference to one group of embodiment being applied to wave filter in the filter coefficient prestored many groups.But, equally likely expect that performance calculates required filter coefficient based on the resonance frequency detected and any other.

In one embodiment of the invention, such as, when signal processing circuit 24 being included in first in the comparatively large equipment comprising loudspeaker 20,22 and loudspeaker 28, or this calibration process is performed when this device first is powered.

In addition, notice, the resonance frequency of loudspeaker can change with temperature, such as, increases and change along with loudspeaker voice coil temperature because of the use of equipment.Therefore, advantageously, in the use of this equipment or perform this calibration after being used for a while.

If expect to perform this calibration when this equipment is in use, then by loudspeaker 28(such as when this equipment is mobile phone in call) useful signal (usefulsignal) (that is, wanted signal and noise-cancelling signal sum) can test signal be used as.

Feedback method based on, the loudspeaker being placed on loudspeaker dead ahead being formed at the use in the chamber between ear and ear casing inside or between ear and mobile phone.Derive from micropkonic signal to be coupled via feedback loop (inverting amplifier) and to get back to loudspeaker, to make it form a servo-drive system, in this servo-drive system, loudspeaker is always attempted to produce zero sound pressure level at loudspeaker place.

Figure 11 show as with reference to Fig. 8 describe, be implemented in feedback system according to the embodiment of signal processing circuit of the present invention.

This feedback system comprises the loudspeaker 120 generally within loudspeaker 128 front.Loudspeaker 120 detects the output of loudspeaker 128, and the signal detected is presented back via amplifier 141 and analog to digital converter 142.Expect that sound signal is fed into this treatment circuit via input end 140.In subtraction element 188, from this expectation sound signal, deduct this by the signal presented back, to make the output of subtraction element 188 substantially represent neighbourhood noise, that is, expect that sound signal is eliminated substantially.

After this, this treatment circuit is similar to the treatment circuit in the feedforward system described with reference to Fig. 8 substantially.The output of subtraction element 188 is fed into adaptive digital filter 144, and the signal through filtering is applied to can adaptation device 146.

Thus, the filtering applied by wave filter 144 and gain apparatus 146 and horizontal adjustment are intended to generate the noise-cancelling signal that the neighbourhood noise that allows to detect is eliminated.

As described above, this noise-cancelling signal is produced by adaptive digital filter 144 and adaptive gain device 146.These are controlled by a control signal, this control signal be by the signal exported from subtraction element 188 is applied to reduce digital sampling rate decimator 152, be then applied to microprocessor 154 and generate.

In this shown embodiment of the present invention, sef-adapting filter 144 is made up of the first filter stage 180 of fixing iir filter 180 form and the second filter stage of self-adaptation Hi-pass filter 182 form.

Microprocessor 154 generates a control signal, and this control signal is applied to self-adaptation Hi-pass filter 182 to adjust its corner frequency.In the use of this noise canceling system, microprocessor 54 generates this control signal on self-adaptation basis, and the performance of wave filter 144 can be adjusted based on the character of the noise signal detected.

But the present invention is equally applicable to the system with fixing wave filter 144.In this linguistic context, word " is fixed " and is meant, and the characteristic of this wave filter is not adjusted based on the noise signal detected.

But, the characteristic of wave filter 144 can be adjusted at calibration phase, when this calibration phase such as can occur in and manufacture this system, or when first this system being integrated in off-the-shelf hardware together with loudspeaker 120 and loudspeaker 128, or when this system is powered, or occur in other erratic mode.

More specifically, by being sent to wave filter 180 by replacing filter coefficient from a group in the many groups coefficient be stored in storer 190, can in the characteristic of the fixing iir filter 180 of this calibration phase adjustment.

Further, can similarly in the gain that the adjustment of this calibration phase is applied by variable gain component 146.Or, by adjusting the characteristic of fixing iir filter 180 aptly, the change of this gain can be realized at this calibration phase.

Like this, just can for the concrete equipment treating to use together to optimize this signal processing circuit.

Microprocessor 154 also generates a test signal, as described above, and this test signal is outputted to adding element 150, and here this test signal is added into the signal exported from adding element 148.Then, composite signal is output to digital to analog converter 152, and is exported by loudspeaker 128.

Figure 12 illustrates in greater detail another embodiment of signal processing circuit 24.Input end 40 is connected to receive---such as direct receive from loudspeaker 20,22---represents the noise signal of neighbourhood noise.This noise signal is imported into analog to digital converter (ADC) 42, and is converted into digital noise signal.This digital noise signal is imported into wave filter 44, and wave filter 44 exports the signal through filtering.Wave filter 44 can be for any wave filter from the ambient noise signal generted noise erasure signal detected, that is, wave filter 44 generates the reverse signal of the neighbourhood noise detected substantially.Such as, wave filter 44 can be adaptive or non-self-adapting, as obvious for those of ordinary skill in the art.

Signal through filtering is output to variable gain module 46.The control of variable gain module 46 will be described after a while.But usually, variable gain module 46 applies gain, to generate the noise-cancelling signal eliminating the neighbourhood noise detected more accurately to the signal through filtering.

Signal processor 24 also comprises the input end 48 for receiving voice or other wanted signal, as described above.This voice signal is imported into ADC 50, and here this voice signal is converted into audio digital signals.Or this voice signal can be received in digital form, and be applied directly to signal processor 24.Then, this audio digital signals is added to the noise-cancelling signal exported from variable gain module 46 in adding element 52.Then, composite signal is outputted to loudspeaker 28 by from signal processor 24.

According to the present invention, this digital noise signal and this audio digital signals are all imported into signal to noise ratio (snr) module 54.SNR module 54 determines the relation between the level of this voice signal and the level of this noise signal, and exports control signal according to determined relation to variable gain module 46.In one embodiment, SNR module 54 detects the ratio of this voice signal and this noise signal, and exports control signal according to the ratio output detected to variable gain module 46.

Term " level " (signal etc.) are used to the value describing signal in this article.This value can be the amplitude of this signal, or the amplitude of the envelope of this signal.Further, this value can be determined by (instantaneously) instantaneously, or is averaging over a period.

Inventor has realized that in the environment such as such as crowded region or concert, the user of noise canceling system 10 always wants this system more to press close to its ear in the environment that neighbourhood noise is high.Such as, if this noise canceling system is implemented in the phone, then this phone may more be pressed close to its ear to listen to the sound of caller better by user.

But the effect done like this makes loudspeaker 28 more press close to ear, thus add the coupling between loudspeaker 28 and ear, that is, the constant level output from loudspeaker 28 will seem more loud concerning user.Further, the coupling between surrounding environment and ear reduces possibly.Such as, when phone, this may be because phone defines and seals more closely around ear, thus has more effectively intercepted neighbourhood noise.

When object be make noise-cancelling signal and neighbourhood noise equivalent and contrary (equal andopposite) time, by making the volume of noise-cancelling signal increase relative to the volume of neighbourhood noise, above-mentioned two kinds of effects have the effect reducing the validity that noise is eliminated.That is, the neighbourhood noise that user hears will be quieter, and noise-cancelling signal will be more loud.Therefore, disagree with intuition, make system 10 actually reduce closer to ear the ability that user listens to voice signal, because this noise is eliminated become more ineffective.

According to the present invention, when system 10 is more pressed close to its ear by user, the gain being applied to this noise-cancelling signal is reduced, to offset above-mentioned effect.Relation between noise signal and voice signal is used to determine when user is in and likely system 10 is more pressed close in the environment of its ear, then reduces this gain.

Such as, in a noisy environment, SNR will be low, and therefore SNR can be used to determine the level to be applied to the gain in gain module 46.In one embodiment, this gain can change continuously along with the SNR detected.In an alternative embodiment, SNR can be compared with a threshold value, and when SNR drops to below this threshold value stepwise (insteps) reduce this gain.In another alternate embodiment, can only when SNR drop to below this threshold value, along with this SNR changes this gain smoothly.

Figure 13 shows the gain of an embodiment and SNR and to reverse the signal chart of relation of thing.As can be seen, when SNR drops to below threshold value SNR0, this gain is reduced smoothly.

Be favourable with comparing of threshold value, unless because when neighbourhood noise is a special problem, system 10 can not may more be pressed close to its ear by user.Therefore, this threshold value can be provided so that gain only reduces when low SNR value.

According to another embodiment, signal processor 24 can comprise slope control module (not shown).This slope control module controls the gain applied in variable gain module 46, does not change rapidly to make this gain.Such as, when system 10 is implemented in the mobile phone, the distance between loudspeaker 28 and ear may be considerable and promptly change.In the case, preferably, the gain being applied to noise-cancelling signal does not change rapidly yet, because this may cause rapid fluctuation, thus stimulates to user.

Under the prerequisite of scope not departing from the claim appended by this instructions, various amendment can be carried out to above-described embodiment.Such as, audio digital signals and/or digital noise signal can be directly inputted to signal processor 28, and in the case, signal processor 28 will not comprise ADC 42,50.Further, SNR module 54 can receive noise signal and the voice signal of analog form, but not digital signal.

For those of ordinary skill in the art it will be clear that, this realization can take the one in several hardware or software form, and the intent of the present invention covers that all these are multi-form.

Can be used in much equipment according to noise canceling system of the present invention, as those of ordinary skill in the art will recognize.Such as, they can be used in mobile phone, head phone, earphone, headset etc.

In addition, it will be appreciated that each aspect of the present invention is applicable to the double any equipment comprising loudspeaker and loudspeaker.Such as, in such devices, the present invention may be used for (the first estimate) according to a preliminary estimate that provide one of loudspeaker and loudspeaker or both sensitivity.The example of such equipment comprises the magnetic recording/reproducing equipment of audio/video, as recording (dictation) equipment, video camera etc.

Those of ordinary skill will be recognized, said apparatus and method can be presented as processor control routine, such as at mounting medium in---such as CD, CD-ROM or DVD-ROM, programmable memory be ROM (read-only memory) (firmware) such as---, or in data carrier on---such as light or electrical signal carrier---.For many application, embodiment of the present invention will be implemented in DSP(digital signal processor), ASIC(special IC) or FPGA(field programmable gate array) on.Thus, this code can comprise conventional program code or microcode (microcode), or such as, for setting up or the code of control ASIC or FPGA.The code that this code also can comprise for dynamically configuring reconfigurable devices---such as reprogrammable logic gate array---.Similarly, this code can comprise for hardware description language---such as Verilog TM or VHDL(Very High Speed Integrated Circuit (VHSIC) hardware description language) code.As skilled in the art will be aware of, this code can be distributed between multiple coupling units of intercoming mutually.In due course, these embodiments also can be used in the scene of running on and can (weight) programmable analogue array or similar devices realize with the code configuring analog/digital hardware.

It should be noted that above-mentioned embodiment is unrestricted the present invention in explanation, and under the prerequisite of scope not departing from claims, those of ordinary skill in the art can design many alternate embodiment.Word " comprises " existence of the element listed by not getting rid of in claim or the element beyond step or step, " one " (" a " or " an ") does not get rid of multiple, and single processor or other unit can realize the function of the some unit described in claim.Any reference number in claim should not be interpreted as the scope limiting claim.

Claims

1. noise canceling system, comprising:

For receiving the input end of digital signal;

Digital filter, has at least high pass filter characteristic, for receiving described digital signal and generating filter output signal;

Wave filter emulator, for receiving described digital signal, and forms the representative of described filter output signal; And

Amplitude detector, the amplitude for the described representative based on described filter output signal generates detection signal;

Wherein said detection signal is applied to described digital filter to control the cutoff frequency of described digital filter,

Wherein said digital signal has the first sampling rate, and described noise canceling system also comprises:

Decimator, for generation of selection input signal, described selection input signal has the second sampling rate lower than described first sampling rate; And

Wherein said selection input signal is applied to described wave filter emulator.

2. noise canceling system according to claim 1, also comprises:

Source input end, for receiving wanted signal; And

Totalizer, for the formation of the described representative sum of described wanted signal and described filter output signal; And

Wherein said amplitude detector is based on described and generate described detection signal with comparing of threshold value.

3. noise canceling system according to claim 2, wherein said amplitude detector based on described and absolute value generate described detection signal with comparing of threshold value.

4., according to described noise canceling system arbitrary in claims 1 to 3, wherein said noise canceling system is feedforward system.

5., according to described noise canceling system arbitrary in claims 1 to 3, wherein said noise canceling system is feedback system.

6. integrated circuit, comprising:

According to described noise canceling system arbitrary in claim 1 to 5.

7. mobile phone, comprising:

Integrated circuit according to claim 6.

8. a pair head phone, comprising:

Integrated circuit according to claim 6.

9. a pair earphone, comprising:

Integrated circuit according to claim 6.

10. headset, comprising:

Integrated circuit according to claim 6.

11. controls are used for the method for the wave filter of noise canceling system, comprising:

Receive digital signal, described digital signal has the first sampling rate;

In digital filter, filtering is carried out to described digital signal, to generate filter output signal;

Apply the emulation of the received digital signal through selecting to described digital filter, to generate simulation data signal, the described received digital signal through selection has the second sampling rate lower than described first sampling rate;

Detect the amplitude of described simulation data signal;

Detection signal is generated based on the described amplitude detected; And

Described detection signal is applied to described digital filter, to control the cutoff frequency of described digital filter.

12. methods according to claim 11, wherein said detecting step also comprises:

Receive wanted signal; And

Form the amplitude sum of described simulation data signal and described wanted signal; And

Wherein said detection signal is based on described and generate with comparing of threshold value.

13. noise canceling systems, comprising:

For the input end of digital signal, this digital signal has the first sampling rate;

Digital filter, comprises fixed filters and self-adaptation Hi-pass filter;

Decimator, for receiving the digital signal of input and generating selection digital signal;

Wave filter emulator, for receiving described selection digital signal; And,

Control circuit, for generating at least one control signal based on the output of described wave filter emulator, at least one control signal wherein said is applied to described sef-adapting filter to control the cutoff frequency of described sef-adapting filter.

14. noise canceling systems according to claim 13, also comprise:

Adaptive gain element;

Wherein said control circuit is suitable for generation second control signal, and described second control signal is used for being applied to described adaptive gain element, to control the gain of described adaptive gain element.

15. noise canceling systems according to claim 13 or 14, wherein said fixed filters is iir filter.

16. noise canceling systems according to claim 13 or 14, also comprise:

Decimator, for receiving described digital signal, and to generate selection signal lower than the second sampling rate of described first sampling rate, wherein said selection signal is applied to described control circuit.

17. noise canceling systems according to claim 13 or 14, also comprise:

For receiving the second input end of wanted signal, described second input end is connected to described control circuit, and described control circuit is suitable for also generating control signal based on input wanted signal.

18. noise canceling systems according to claim 13 or 14, wherein said control circuit is suitable for generating at least one control signal described based on the supplied with digital signal of non-filtered.

19. noise canceling systems according to claim 13 or 14, wherein said control circuit is suitable for generating at least one control signal described based on the output of described fixed filters.

20. noise canceling systems according to claim 13 or 14, wherein said control circuit is suitable for generating at least one control signal described based on the output of described self-adaptation Hi-pass filter.

21. noise canceling systems according to claim 13 or 14, wherein said noise canceling system is feedforward system.

22. noise canceling systems according to claim 13 or 14, wherein said noise canceling system is feedback system.

23. integrated circuit, comprising:

According to claim 13 to described noise canceling system arbitrary in 22.

24. mobile phones, comprising:

Integrated circuit according to claim 23.

25. a pair head phone, comprising:

Integrated circuit according to claim 23.

26. a pair earphone, comprising:

Integrated circuit according to claim 23.

27. headsets, comprising:

Integrated circuit according to claim 23.

28. controls are used for the method for the wave filter of noise canceling system, comprising:

Receive digital signal, described digital signal has the first sampling rate;

With fixed filters, filtering is carried out to described digital signal;

With self-adaptation Hi-pass filter, filtering is carried out to described digital signal;

Select to generate selection digital signal to received digital signal; And

Apply described selection digital signal to wave filter emulator;

Wherein said self-adaptation Hi-pass filter is suitable for the cutoff frequency controlling described self-adaptation Hi-pass filter based on the output of described wave filter emulator.

29. methods according to claim 28, also comprise:

Generate selection signal from described supplied with digital signal, described selection signal has the second sampling rate lower than described first sampling rate;

Wherein said sef-adapting filter is adapted based on described selection signal.